--- a/misc/winutils/include/SDL_audio.h Wed May 16 18:22:28 2018 +0200
+++ /dev/null Thu Jan 01 00:00:00 1970 +0000
@@ -1,284 +0,0 @@
-/*
- SDL - Simple DirectMedia Layer
- Copyright (C) 1997-2012 Sam Lantinga
-
- This library is free software; you can redistribute it and/or
- modify it under the terms of the GNU Lesser General Public
- License as published by the Free Software Foundation; either
- version 2.1 of the License, or (at your option) any later version.
-
- This library is distributed in the hope that it will be useful,
- but WITHOUT ANY WARRANTY; without even the implied warranty of
- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- Lesser General Public License for more details.
-
- You should have received a copy of the GNU Lesser General Public
- License along with this library; if not, write to the Free Software
- Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
-
- Sam Lantinga
- slouken@libsdl.org
-*/
-
-/**
- * @file SDL_audio.h
- * Access to the raw audio mixing buffer for the SDL library
- */
-
-#ifndef _SDL_audio_h
-#define _SDL_audio_h
-
-#include "SDL_stdinc.h"
-#include "SDL_error.h"
-#include "SDL_endian.h"
-#include "SDL_mutex.h"
-#include "SDL_thread.h"
-#include "SDL_rwops.h"
-
-#include "begin_code.h"
-/* Set up for C function definitions, even when using C++ */
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/**
- * When filling in the desired audio spec structure,
- * - 'desired->freq' should be the desired audio frequency in samples-per-second.
- * - 'desired->format' should be the desired audio format.
- * - 'desired->samples' is the desired size of the audio buffer, in samples.
- * This number should be a power of two, and may be adjusted by the audio
- * driver to a value more suitable for the hardware. Good values seem to
- * range between 512 and 8096 inclusive, depending on the application and
- * CPU speed. Smaller values yield faster response time, but can lead
- * to underflow if the application is doing heavy processing and cannot
- * fill the audio buffer in time. A stereo sample consists of both right
- * and left channels in LR ordering.
- * Note that the number of samples is directly related to time by the
- * following formula: ms = (samples*1000)/freq
- * - 'desired->size' is the size in bytes of the audio buffer, and is
- * calculated by SDL_OpenAudio().
- * - 'desired->silence' is the value used to set the buffer to silence,
- * and is calculated by SDL_OpenAudio().
- * - 'desired->callback' should be set to a function that will be called
- * when the audio device is ready for more data. It is passed a pointer
- * to the audio buffer, and the length in bytes of the audio buffer.
- * This function usually runs in a separate thread, and so you should
- * protect data structures that it accesses by calling SDL_LockAudio()
- * and SDL_UnlockAudio() in your code.
- * - 'desired->userdata' is passed as the first parameter to your callback
- * function.
- *
- * @note The calculated values in this structure are calculated by SDL_OpenAudio()
- *
- */
-typedef struct SDL_AudioSpec {
- int freq; /**< DSP frequency -- samples per second */
- Uint16 format; /**< Audio data format */
- Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */
- Uint8 silence; /**< Audio buffer silence value (calculated) */
- Uint16 samples; /**< Audio buffer size in samples (power of 2) */
- Uint16 padding; /**< Necessary for some compile environments */
- Uint32 size; /**< Audio buffer size in bytes (calculated) */
- /**
- * This function is called when the audio device needs more data.
- *
- * @param[out] stream A pointer to the audio data buffer
- * @param[in] len The length of the audio buffer in bytes.
- *
- * Once the callback returns, the buffer will no longer be valid.
- * Stereo samples are stored in a LRLRLR ordering.
- */
- void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
- void *userdata;
-} SDL_AudioSpec;
-
-/**
- * @name Audio format flags
- * defaults to LSB byte order
- */
-/*@{*/
-#define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
-#define AUDIO_S8 0x8008 /**< Signed 8-bit samples */
-#define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */
-#define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */
-#define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */
-#define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */
-#define AUDIO_U16 AUDIO_U16LSB
-#define AUDIO_S16 AUDIO_S16LSB
-
-/**
- * @name Native audio byte ordering
- */
-/*@{*/
-#if SDL_BYTEORDER == SDL_LIL_ENDIAN
-#define AUDIO_U16SYS AUDIO_U16LSB
-#define AUDIO_S16SYS AUDIO_S16LSB
-#else
-#define AUDIO_U16SYS AUDIO_U16MSB
-#define AUDIO_S16SYS AUDIO_S16MSB
-#endif
-/*@}*/
-
-/*@}*/
-
-
-/** A structure to hold a set of audio conversion filters and buffers */
-typedef struct SDL_AudioCVT {
- int needed; /**< Set to 1 if conversion possible */
- Uint16 src_format; /**< Source audio format */
- Uint16 dst_format; /**< Target audio format */
- double rate_incr; /**< Rate conversion increment */
- Uint8 *buf; /**< Buffer to hold entire audio data */
- int len; /**< Length of original audio buffer */
- int len_cvt; /**< Length of converted audio buffer */
- int len_mult; /**< buffer must be len*len_mult big */
- double len_ratio; /**< Given len, final size is len*len_ratio */
- void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
- int filter_index; /**< Current audio conversion function */
-} SDL_AudioCVT;
-
-
-/* Function prototypes */
-
-/**
- * @name Audio Init and Quit
- * These functions are used internally, and should not be used unless you
- * have a specific need to specify the audio driver you want to use.
- * You should normally use SDL_Init() or SDL_InitSubSystem().
- */
-/*@{*/
-extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
-extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
-/*@}*/
-
-/**
- * This function fills the given character buffer with the name of the
- * current audio driver, and returns a pointer to it if the audio driver has
- * been initialized. It returns NULL if no driver has been initialized.
- */
-extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen);
-
-/**
- * This function opens the audio device with the desired parameters, and
- * returns 0 if successful, placing the actual hardware parameters in the
- * structure pointed to by 'obtained'. If 'obtained' is NULL, the audio
- * data passed to the callback function will be guaranteed to be in the
- * requested format, and will be automatically converted to the hardware
- * audio format if necessary. This function returns -1 if it failed
- * to open the audio device, or couldn't set up the audio thread.
- *
- * The audio device starts out playing silence when it's opened, and should
- * be enabled for playing by calling SDL_PauseAudio(0) when you are ready
- * for your audio callback function to be called. Since the audio driver
- * may modify the requested size of the audio buffer, you should allocate
- * any local mixing buffers after you open the audio device.
- *
- * @sa SDL_AudioSpec
- */
-extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);
-
-typedef enum {
- SDL_AUDIO_STOPPED = 0,
- SDL_AUDIO_PLAYING,
- SDL_AUDIO_PAUSED
-} SDL_audiostatus;
-
-/** Get the current audio state */
-extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);
-
-/**
- * This function pauses and unpauses the audio callback processing.
- * It should be called with a parameter of 0 after opening the audio
- * device to start playing sound. This is so you can safely initialize
- * data for your callback function after opening the audio device.
- * Silence will be written to the audio device during the pause.
- */
-extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
-
-/**
- * This function loads a WAVE from the data source, automatically freeing
- * that source if 'freesrc' is non-zero. For example, to load a WAVE file,
- * you could do:
- * @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode
- *
- * If this function succeeds, it returns the given SDL_AudioSpec,
- * filled with the audio data format of the wave data, and sets
- * 'audio_buf' to a malloc()'d buffer containing the audio data,
- * and sets 'audio_len' to the length of that audio buffer, in bytes.
- * You need to free the audio buffer with SDL_FreeWAV() when you are
- * done with it.
- *
- * This function returns NULL and sets the SDL error message if the
- * wave file cannot be opened, uses an unknown data format, or is
- * corrupt. Currently raw and MS-ADPCM WAVE files are supported.
- */
-extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
-
-/** Compatibility convenience function -- loads a WAV from a file */
-#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
- SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
-
-/**
- * This function frees data previously allocated with SDL_LoadWAV_RW()
- */
-extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf);
-
-/**
- * This function takes a source format and rate and a destination format
- * and rate, and initializes the 'cvt' structure with information needed
- * by SDL_ConvertAudio() to convert a buffer of audio data from one format
- * to the other.
- *
- * @return This function returns 0, or -1 if there was an error.
- */
-extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
- Uint16 src_format, Uint8 src_channels, int src_rate,
- Uint16 dst_format, Uint8 dst_channels, int dst_rate);
-
-/**
- * Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),
- * created an audio buffer cvt->buf, and filled it with cvt->len bytes of
- * audio data in the source format, this function will convert it in-place
- * to the desired format.
- * The data conversion may expand the size of the audio data, so the buffer
- * cvt->buf should be allocated after the cvt structure is initialized by
- * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.
- */
-extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt);
-
-
-#define SDL_MIX_MAXVOLUME 128
-/**
- * This takes two audio buffers of the playing audio format and mixes
- * them, performing addition, volume adjustment, and overflow clipping.
- * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
- * for full audio volume. Note this does not change hardware volume.
- * This is provided for convenience -- you can mix your own audio data.
- */
-extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume);
-
-/**
- * @name Audio Locks
- * The lock manipulated by these functions protects the callback function.
- * During a LockAudio/UnlockAudio pair, you can be guaranteed that the
- * callback function is not running. Do not call these from the callback
- * function or you will cause deadlock.
- */
-/*@{*/
-extern DECLSPEC void SDLCALL SDL_LockAudio(void);
-extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
-/*@}*/
-
-/**
- * This function shuts down audio processing and closes the audio device.
- */
-extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
-
-
-/* Ends C function definitions when using C++ */
-#ifdef __cplusplus
-}
-#endif
-#include "close_code.h"
-
-#endif /* _SDL_audio_h */